1. Field of Invention
This invention relates generally to communications techniques, and more particularly to systems and methods for facilitating simultaneous transmission of a multiplicity of channels of information over a common transmission medium.
2. Status of Prior Art
Communication: The first copper-wire communication system was only capable of carrying one message per wire. Communications companies soon realized that, in order to enlarge their capacity to carry messages, they would have to devise ways to transmit several messages simultaneously over a single wire, for the cost of installing additional lines to accommodate increased demand was high. Companies that could reduce costs by putting more and more information over a single line, would have a competitive advantage. Discoveries in transmission techniques enabled more than one message to be transmitted per line, thereby paving the way for our telegraph and telephone industry to become viable commercial enterprises. The challenge of maximizing bandwidth and increasing line capacity was present from the very beginning of telecommunications technology, and is still with us today.
Presently, telecommunication networks provide the primary means for conveying voice and data traffic between source and destination. But existing telecommunication networks cannot handle the ever-increasing demand for transmission capacity. Rising population, lower telephone rates and increased data traffic over the Internet, all underscore the need to increase network capacity. As more and more bandwidth becomes available, higher bandwidth applications are quickly developed, such as higher resolution web pages and video-on-demand, which once again heightens the demand for increased bandwidth.
One way to satisfy an increasing demand for bandwidth is by installing additional transmission lines or by placing additional satellites in the sky. Both solutions are expensive and dictate substantial investments. Yet, even satellite solutions have limitations, for there is only a limited number of satellites that can be placed in geostationary orbit in the Clarke belt. After all, the Clarke belt is the only location where satellites, when viewed from the Earth's surface, remain substantially stationary, thereby permitting the use of fix-mounted dish antennas. Wireless systems, where the available radio spectrum is limited, also rely on bandwidth utilization or compression methods to expand the capacity of the system. To remain competitive, network service providers must endeavor to preserve the functionality of their existing networks, yet still be able to accommodate the increasing bandwidth demand to handle voice, data, and video transmission.
In conventional analog transmission, voice energy acts to compress the carbon granules in a microphone, thereby varying the microphones resistance to electrical current. Then the varying current, which varies in a manner analogous to the acoustical vibrations of the speaker's voice, is used to energize an electromagnet, actuating a diaphragm which vibrates to reproduce the original voice. Digital transmission adds several steps to this transformation, for the voice is converted to an electrical current pattern whose varying amplitude is measured thousands of times per second. These measurements are encoded as binary numbers, consisting of “0” and “1”s.
Unlike analog transmission which conveys the sound as a continuous wave form, in digital transmission, binary numbers are transmitted in representational encoding schemes. Binary digits or bits, may be transmitted singly, as discrete, on-off or zero/non-zero current pulses, or in groups as simultaneous pulses at different frequencies. At the receiving end, the bit stream is interpreted and the numbers reconstituted to modulate a current which drives a speaker. This method is “digital” because it entails conversion of an analog signal to numbers, and the transmission of digits in symbolic form.
Compression: There are several known methods which make possible the transmission of information with diminished bandwidth requirements. The most widely employed method relating to “compression” uses mathematical algorithms and dictionary tables to manipulate and “point” digital signals in such a way that each transmission channel uses less bandwidth to carry recognizable information. Compression is achieved by building a predictive model of the waveform, removing unnecessary elements, and reconstructing the waveform from the remaining elements.
When converting an analog signal into digital form, accurate conversion requires at least twice as many measurements (samples per second), as the highest frequency in the signal. The human voice generates sound frequencies in a range of 20 to 4,000 Hz. Hence, an ideal digital voice circuit, accepting an input in the range of 0–4,000 Hz, must sample this signal 8,000 times per second. Each sample is represented by 8 bits of data, and a single voice circuit, referred to as DSO, “digital signal level zero”, carries 64,000 (8,000×8) bits of data.
Compression methods are based on reducing the number of bits required to convey a human voice or other data transmission. Currently utilized compression algorithms can produce acceptable voice quality using less than 64 kbs by eliminating frequencies not necessary for voice intelligibility, particularly those below 300 Hz and those above 3,300 Hz, and emphasizing the frequencies in the 1,000 Hz range that carry most of the voice energy. Methods that drop an excessive amount of input signal tend to frustrate high-speed tonal data transmission schemes employed by modems and faxes. Currently-employed compression algorithms and equipment are able to transmit acceptable voice quality with a compression ratio of 8:1, using 8,000 bps per channel.
With these compression methods, one channel can be made to carry eight voice conversations or eight fax transmission over a line that originally was able to carry only one voice conversation. Higher compression methods which transmit voice and data over a circuit using less than 8,000 bps, suffer from increasing degradation of voice quality and “loss,” whereby at the receiving end of the line the voice in its original form is not clearly heard. Although new methods and algorithms may be employed to allow for clear voice transmission using less than 8,000 bps, there are appreciable limitations to these methods. All compression methods using algorithms suffer from greater and greater “loss” as compression ratios increase. Fax and video transmission that are more sensitive to bandwidth degradations are more limited in their acceptable compression ratios.
While the main advantage of digital compression is that it increases network efficiency, it can in some situations reduce it. Users of compression technology must ensure that their chosen compression method has the ability to transmit compressed data at the full capacity of the transmission lines. If not, consideration must be given to downgrading the speed of the transmission lines and sacrificing some of the throughput. Furthermore, the amount of time the computer spends compressing and decompressing the data can reduce efficiency.
Multiplexing: The most common form of telecommunications service is T-1 protocol. T-1 uses a form of multiplexing in which 24 voice or data channels, each with 8,000 bps, can simultaneously exist on one pair of twisted copper wires. The total bandwidth capacity of T-1 is 1.544 Mbps. Compression methods are used in conjunction with T-1 and other transmission protocols to maximize bandwidth. Common compression systems using a ratio of 8:1, can carry 192 simultaneous voice or data channels (24×8) over a T-1 line.
Network service providers employ methods for increasing bandwidth through the utilization of compression and multiplexing, the most common multiplexing scheme in the United States being the T-1 protocol. Conversations or digital information carried on each T-1 line or channel is rendered unique, and transmitted with other channels over a common medium by multiplexing.
An early method used by phone companies to render channels unique, is Frequency Division Multiplexing (FDM). In FDM, each of the 24 channels are rendered distinct by having each channel assigned to a frequency band. (For example, line 1 would use the frequency band of 0Hz–4,000 Hz, line 2 would use the 4,000 Hz–8,000 Hz band, etc.) But this method is best suited for analog signals which are subject to degradation and noise interference, and is therefore not commonly used at present. More common techniques are Time Division Multiplexing (TDM) and Statistical Multiplexing (STDM), often called “Packet switching.” In TDM, each of the 24 channels (or lines) are rendered distinct by having each channel assigned to a particular, non-overlapping time slot. Frames of 24 time slots are transmitted, in which Channel 1 gets the first time slot in the frame, Channel 2 gets the second time slot and so on. STDM works in a similar manner to TDM, assigning channels on the basis of time division. But STDM takes advantage of statistical fluctuations, and instead of automatically assigning each channel to a time slot, STDM assigns only active channels to time slots. Hence, instead of transmitting channels in sequential order (1, 2, 3, 4, 5, 6) as in TDM, STDM only assigns time slots to channels that are being used, e.g., 1, 3, 1, 5, 1, 6 etc. This method creates higher bandwidth utilization than TDM.